1. Field of the Invention
The present invention relates stethoscopes and in particular to coding and digitally transmitting auscultation sounds to a remote location where the signals are decoded and converted back to analog sounds for listening.
2. Description of Related Art
When a clinician examines a patient, the clinician typically will check the patient's vital signs. Key among the vital signs typically checked are the cardiovascular (heart) and respiratory (lung) sounds of the patient. Sounds coming from the body are referred to as auscultation sounds and a stethoscope is the medical device used to listen to a patient's auscultation sounds. A typical stethoscope will have a bell and diaphragm capability, where the bell position slightly enhances the lower frequency sounds of the beating heart and the diaphragm position is better at passing the higher frequencies, such as the breath sounds of the lungs.
The clinician will place the chest piece of the stethoscope to various spots on the patient's front and back depending on which organ is being monitored. The clinician will also use the bell/diaphragm capability as appropriate to enhance the auscultation exam. While a clinician may wish to see the patient and where the chest piece is being placed, it is not essential that the clinician hold the chest piece. That is, someone else, including the patient can position and hold the chest piece following the instructions of the clinician. As long as there is a video/audio (or at least an audio) connection between the clinician and patient and a data communications channel to pass auscultation sounds, the physician and patient don't have to be physically in the same location. The ability to perform certain medical functions on a patient at a remote location is generally referred to as telemedicine. Having a remote telephonic stethoscope system is essential in performing medical tests that require auscultation in telemedicine.
The output of a typical amplifying stethoscope can be digitized and send over a digital communications channel to a receiving stethoscope unit that converts the digitized signal back to analog to allow a clinician to listen to the sounds. The techniques and equipment available in the prior art to accomplish this produce a digital signal with a data rate that significantly limits the frequency range of the auscultation signal that may be passed over normal telephone lines; hence, many systems are thus restricted to usage between facilities that have broadband communications channels. Currently, most homes do not have broadband communications service. However, nearly all homes have basic telephone service so a remote telephonic stethoscope system that uses a bit rate low enough to be used over a normal telephone line is highly advantageous. Some remote telephonic stethoscope system can accomplish this in a store-and-forward method that approaches real-time operation. That is, they store a brief period of auscultation sounds and then pause in the monitoring of new sounds while they send the stored sounds. This is not real-time and is awkward for the clinician and patient.
Typical remote telephonic stethoscope systems use a relatively large bandwidth, generally in the range of 32 Kb/s to 64 Kb/s. The best state-of-the-art system achieves a modest auscultation bandwidth using 9.6 Kb/s, uses special techniques for error handling and achieves a bandwidth of 30 Hz to 500 Hz.
The prior art includes a remote telephonic stethoscope system that uses Pulse Code Modulation (PCM) and Adaptive Differential PCM (ADPCM) coding and a repeated byte error handling circuit to achieve a low bit rate that passes part of the key auscultation frequency band of interest and operates in real-time, as shown in U.S. Pat. No. 5,841,846. The commercial implementation of this patent also includes in the remote stethoscope unit a monitor port so that the local physician with the patient can listen as well as the remote physician.
The conversion of an analog auscultation signal to a digital auscultation signal (A/D conversion) involves sampling the analog signal periodically and quantizing the samples. How often the analog signal is sampled is dependent on the highest frequency component that is to be passed. According to the Nyquist criteria, it is necessary to sample an analog signal at least twice as often as the highest frequency to be transmitted. For example, to pass a signal up to 3,400 Hz (cycles per second) requires sampling at least 6,800 times a second. While an analog signal is continuous, its digital counterpart is not. Matching the analog sample to its nearest digital equivalent is called quantization. Linear coding typically requires 10–12 bits for each sample and is the easiest and least efficient. PCM provides better efficiency and achieves similar quality with only 8 bits per sample by using a technique called companding where greater sensitivity is given to low volume sounds by assigning relatively more digital values than for high volume sounds. Because the typical signals are band limited in some way, the amplitude difference between two adjacent samples is much smaller than the total possible amplitude range. More sophisticated schemes take advantage of that and produce even greater bandwidth efficiency. Adaptive Differential PCM (ADPCM) yields nearly the same quality as PCM but produces only four bits per sample.
Adaptive delta modulation (ADM), although a known coding scheme, has not been used an electronic stethoscope system. ADM is a sophisticated digital coding method that uses an adaptive differential quantization technique based on the differences between three or four adjacent samples, but only produces one bit for each quantization computation. By looking at multiple samples, both the rate of change as well as the change in amplitude and slope can be used in the computation. Since each computation produces only one bit, rather than four, eight or twelve, there is no need for any synchronization or framing patterns; it is inherently self-synchronizing. A specific implementation of ADM called continuously variable slope delta (CVSD) modulation is used in communications networks where efficiency and robustness against noise is needed.
ADM/CVSD integrated circuits (IC) have been designed for the telephony communications industry. They are tailored for the telephony voice frequency range of 300 Hz to 3,400 Hz and typically include at the encoding side an input filter, an A/D converter and ADM/CVSD algorithm processing and at the decoding side ADM/CVSD algorithm processing, D/A conversion and output filtering.